The present invention is generically directed on a technique for so-called xe2x80x9cbeam formingxe2x80x9d on acoustical signals.
The use of directional acoustical/electrical transducers and especially of such microphones is one of the most efficient ways for improving signal to noise ratio in audio systems. It is known to realise directional microphones by using an array of microphone cells and time delaying and superimposing the output signals of such cells following up the known xe2x80x9cdelay and sumxe2x80x9d technique.
With two omnidirectional microphone cells this known principle is shown in FIG. 1. Two omnidirectional microphones, 1 and 2, are provided with a mutual distance p. The output signal of one of the microphones according to signal A1 is time delayed by the time amount xcfx84, the time delayed signal according to A1xe2x80x2 is superimposed at a superimposing unit 3 to the undelayed output signal A2 of microphone 2. At the output of the superimposing unit 3 there results the output signal Ar with an amplification versus impinging angle xcex8 characteristic, as shown in FIG. 2 for one frequency xcfx89 considered. Thereby, it is customary to select as delay time xcfx84 as the quotient of distance p and velocity of sound c. With this arrangement there results, as shown in FIG. 2, a first order cardoid characteristic. It may be shown that the amplitude of the resulting signal Ar is proportional to the sine of the signal frequency xcfx89 and to the distance p. The maximum gain in target direction (180xc2x0) occurs at the frequency fr=c/(4p) . For a distance p of 12 mm, fr becomes approx. 7 kHz.
By staggering more than one of the FIG. 1 double-cell arrangements and superimposing the resulting signals Ar of the more than one double-cell arrangements, higher order cardoid characteristics may be realised.
In FIG. 3 a known arrangement to realise second order cardoid characteristics according to FIG. 4 is shown. Thereby, a narrower beam can be achieved. The higher the order of the directional microphone arrangement, the higher becomes the directivity index and the gain at fr, but the higher will also be the roll-off for low and high frequencies and the number of unwanted side-lobes. With respect to the definition of the directivity index please refer to speech communication 20 (1996), 229-240, xe2x80x9cMicrophone array systems for hands-free telecommunicationsxe2x80x9d, Garry W. Elko.
In FIG. 5 there is shown the gain versus frequency characteristic of the first and second cardoid characteristics for an impinging angle xcex8=180xc2x0. Therefrom, high and low frequency roll-offs are clearly evident.
Such techniques for beam forming are well-known and have been realised using analogue signal processing, as e.g. shown in the U.S. Pat. Nos. 2,237,298, 4,544,927, 4,703,506, 5,506,908 or using digital signal processing, both in time or in frequency domain, as shown in the EP-A-0 381 498 (time domain) or in the U.S. Pat. No. 5,581,620 (frequency domain).
Beam formings realised with any of these principles has the following drawbacks:
a) The resulting signal is dampened at low frequencies, which results in a bad signal to noise ratio.
b) The directivity index is very sensitive to matching of the individual microphone cells, especially at low frequencies.
c) The distance p between the microphone cells should be large ( greater than 12 mm) for audio range.
d) The frequency band with a high gain in target direction is rather small, as may clearly be seen from FIG. 5.
e) The directivity largely depends upon the number of microphone cells and thus on the complexity of the overall arrangement.
f) As one aims for a high directivity by increasing the number of cells, more unwanted side-lobes are introduced.
Several techniques have been proposes to overcome some of these drawbacks:
In the WO 95/20305 (E. Lindemann) an adaptive noise reduction system for use in binaural hearing aid is proposed. It detects the power of the received signals to separate the desired from unwanted signals.
There is proposed a xe2x80x9cbroad sidexe2x80x9d microphone-cell array, i.e. target direction is perpendicular to the axis from one microphone to the other, in contrary to the arrangement according e.g. to FIG. 1 and the principles of the present invention, which is xe2x80x9cin linexe2x80x9d.
The disclosed apparatus is bulky ( greater than  greater than 5 cm), so that it may not be implemented for one ear hearing aid.
Two equal beam lobes are generated in target and in opposing directions.
In such a hearing aid a connection between the left and right ear system must be present, making the apparatus for hearing aid unhandy. Furthermore, as described by the same author in xe2x80x9cTwo microphone non-linear frequency domain beam former for hearing aid noise reductionxe2x80x9d 1995, IEEE ASSP Workshop on Applications of Signal Processing to Audio and Acoustics, October 15-18, Mohonk, New Paltz, New York, such beam forming is efficient only up to about 2 kHz and leads to distortions of the desired signals.
The U.S. Pat. No. 4,653,102 proposes the use of two directional microphones aimed in target direction and of a third microphone aimed in opposite direction. The signal of the third microphone supposedly only containing noise is used to shape the response of the two primary microphones. This technique obviously has the drawback within reverberating rooms, where the desired signal is reflected on walls, floor, ceiling and furniture and is therefore considered as noise by the system. This technique is further unhandy as making use of at least three microphones.
Attention is further drawn to U.S. Pat. Nos. 5,400,409 and 5,539,859.
As an example of known beam-forming techniques, the U.S. Pat. No. 5,539,859 proposes a technique wherein reception characteristic is logged in on that direction wherefrom the highest energy impinges on a pair of microphones and considered in the sound environment. Principally, all sound impinging from directions other than from highest energy direction is considered as noise and its reception is cancelled.
Thereby, an analogue to digital conversion and subsequent time to frequency domain conversion is performed on the output signals of two microphones. Exploiting the knowledge of the fixed mutual distance between the two microphones, wherefrom phase difference of the impinging signal spectra is dependent, there is determined the mutual phasing and thus impinging direction of highest energy sound signals, i.e. direction of highest energy sound source within the acoustical surrounding. Signals impinging from that direction are amplified by means of inphase shifting and adding similarly to an auto correlation technique, whereby signals from other impinging angles are cancelled as noise.
By such a technique the energy distribution in the sound environment traps the selectivity of reception, and it is not possible to freely select or preselect a maximum reception characteristic, e.g. in direction wherefrom sound is desired to be selectively received, irrespective of its relative energy. One field whereat such selectivity irrespective of energy distribution within the sound surrounding would clearly be advantageous is hearing aid technique.
It is an object of the present invention to provide a method for electronically forming a predetermined characteristic of amplification in dependency of direction from which acoustical signals are received at at least two spaced apart acoustical/electrical transducers and a respective acoustical sensor apparatus, with which only a small number of microphones or microphone cells has to be used and which is thus enabling small and compact directional transducer or microphone realisation. Thereby, the preferred apparatus according to the present invention is a hearing aid apparatus, and especially a one ear hearing aid apparatus.
It is a further object to provide such method and apparatus with good frequency response in the audio band, i.e. between approx. 0.1 and 10 kHz.
Still a further object of the present invention is to provide such method and apparatus which allow high signal to noise ratio realisation without unwanted side-lobes and with easily variable beam form, e.g. for acoustical zooming.
These and other objects are realised by the inventive method, which comprises the steps of repetitively determining from signals dependent from the acoustical signals a respective mutual delay signal according to reception delay at the at least two transducers; subjecting a signal dependent from the output signal of at least one of the at least two transducers to filtering with a filtering transfer characteristic; and of controlling the filtering transfer characteristic in dependency of the mutual delay signal; further exploiting a signal dependent from the output signal of the filtering as electrical reception signal.
To fulfil the above mentioned objects the inventive acoustical sensor apparatus comprises at least two acoustical/electrical transducers, arranged at a predetermined mutual distance in target direction, a time delay detection unit, which has at least two inputs and an output, the inputs thereof being respectively operationally connected to the outputs of the two transducers, whereby the time delay detection unit generates an output signal in dependency of the time delay of acoustical signals, impinging on the at least two spaced apart transducers, preferably a time domain to frequency domain converter unit generating the output signal of said time delay detection unit in frequency domain; a weighing unit with a predetermined weighing characteristic and with an input and with an output, whereby the input thereof is operationally connected to the output of the time delay detection unit and preferably receiving the signal at said output of said time delay detection unit in frequency domain mode; with a filter unit with a controllable transfer characteristic, which has at least one input, a control input and an output and whereat the input is operationally connected to at least one of the outputs of the at least two transducers, preferably via at least one time domain to frequency domain converter, the control input is operationally connected to the output of the weighing unit, the filter unit generating an output signal in dependency of its input signal and of its transfer characteristic which is controlled by the signalxe2x80x94preferably a spectral signalxe2x80x94which is applied to the control input of the filter unit, this weighingxe2x80x94preferably spectral weighingxe2x80x94result signal being dependent from the output signal of the time delay detection unit and the weighing characteristic of the weighing unit.